1. Field of the Invention
The present invention relates to a so-called VoIP (Voice over Internet Protocol) technique which transfers audio (sound or voice) data by using an IP network and a technique which minimizes an audio quality deterioration factor such as a packet loss or jitter which is generated by congestion in a relay router.
2. Description of the Related Art
In recent years, transition from audio communication by STM exchange to audio communication by the VoIP technique using an IP network (IPNW) is rapidly advanced. FIG. 22 is a diagram showing a VoIP network. In FIG. 22, the VoIP network is constituted by a VoIP gateway (VoIP-GW), a CA (Call Agent: also called a “gate keeper”), and a relay router. The VoIP-GW is generally located between a PSTN (Public Switched Telephone Network) and an IPNW. The VoIP-GW transmits an audio packet (voice packet) obtained by packeting PCM (Pulse Code Modulation) audio data from the PSTN to the IPNW side or converts the packet from the IPNW into PCM audio data and transmits the PCM audio data to the PSTN. An RTP/UDP is used as an upper layer of the IP layer of an audio packet. A calling user and a called user are discriminated from each other by using an UDP-Port number. A CA executes call control for outgoing/incoming of calls or the like with the PSTN. As shown in FIG. 22, when an IP telephone is directly accommodated in an IP network, a CA executes call control the IP telephone based on H. 323(In this case, the IP telephone is connected with the IP network via an edge router). In the call control based on H. 323, the CA designates the IP address of the VoIP-GW of a destination, UDP-Port, codec format (e.g., G. 711, G. 723, G729), and the like. On the other hand, the CA controls the VoIP-GW by using, e.g., Megaco (Media Gateway Control). The relay router executes a relaying (forwarding) operation of an audio packet transmitted and received by the VoIP-GW and an edge router and an IP packet of other data.
When audio data is transferred by using an IP network, it is important how to reduce audio deterioration caused by packet loss, jitter, and the like while keeping real-time properties. In a conventional IP network, in general, Differentiated Service (Diff-serv) using IP-ToS (Type of Service) or QoS (Quality of Service) control by an RSVP (Resource Reservation Protocol) for band reservation control is executed.
FIG. 23 is a diagram showing priority control executed by the Diff-serv. In the Diff-serv, a relay router has a plurality of cues to which predetermined priorities (e.g., “high priority”, “intermediate priority”, and “low priority” are assigned. The relay router refers to the values of ToS set in the headers of the received packets and distributes the received packets to the cues according to the priorities corresponding to the ToS values. Thereafter, the relay router executes read control according to the priorities. At this time, a packet set in the “high priority” is preferentially read out and transferred to the next hop. In this manner, the packets having the “high-priority” are restrained being delayed. When the VoIP technique is applied, ToS corresponding to a “high priority” is set in an audio packet such that the audio packet has a priority higher than that of a data packet.
However, the Diff-serv is a technique in order to discriminate an audio packet flow from other data flows on the basis of priorities (for example, a high priority is given to an audio packet, and low priorities are given to the other data packets). The Diff-serv does not guarantee audio flow bands required for telephonic communication (audio communication) every telephonic communication. For this reason, as shown in FIG. 24, when congestion is generated in a flow distributed with high priority, packet loss (wasting or discarding an audio packet caused by an overflow of cues) is generated in each audio flow. Quality deterioration may be caused by mutual influences.
On the other hand, an RSVP reserves resources such as a band and a buffer between ends (end-to-end) through an IPNW to implement QoS control. For this reason, as shown in FIG. 25, a connection is set between ends (end-to-end). Band reservation control executed by an RSVP is executed in all the relay routers which relay a packet transferred on this connection to execute flow discrimination, band monitor, and the like.
However, the VoIP is an application which has a large number of low-band flows and which frequently executes call connection/disconnection. For this reason, when the VoIP is handled by the RSVP, a relay router must frequently execute the RSVP and must execute band monitor or the like based on the RSVP. Therefore, a load on the relay router increases. In addition, if a scale of the network increases and relay routers on an IP connection increase, QoS control by the RSVP increases and becomes complex. As described above, the QoS control executed by the RSVP has a problem in scalability. The QoS control by the RSVP cannot be actually employed by a large-scale network such as a carrier network.